void EigenVectors::GetEigenVectors(DMatrix &egvts, DArray &egvls) const {
EigenValues egvl(input_);
egvl.ComputeEigenValues();
egvls = egvl.GetRealEigenValues();
int size = egvls.size();
InitMatrix(egvts, size);
for(int i = 0; i < size; ++i) {
egvls[i] = RoundToZero(egvls[i]);
FindEigenVector(egvts, egvls[i], i);
}
}
void EigenVectors::FindEigenVector(DMatrix &egvts, double egvl, int ind) const {
DArray egvt, darr;
DMatrix dmat = input_, dmatprev;
double det = 0.;
int i1, i2, j1, size = egvts.size(), rs = size;
InitArray(egvt, size);
darr = egvt;
for(i1 = 0; i1 < size; ++i1) {
dmat[i1][i1] -= egvl;
}
for(i1 = 0; i1 < ind && det == 0.; ++i1) {
for(j1 = 0; j1 < size; ++j1) {
dmat[i1][j1] = egvts[j1][i1];
}
det = ComputeSystem(dmat, darr);
}
if(det == 0.) {
egvt[ind] = 1.;
dmatprev = dmat;
rs = TrimSystem(dmat, darr);
PopulateSystem(dmat, darr, dmatprev, ind);
while(rs > 1) {
det = ComputeSystem(dmat, darr);
if(det != 0.) {
break;
}
rs = TrimSystem(dmat, darr);
}
if(det == 0.) {
darr[0] = (dmat[0][0] == 0.) ? 0. : (darr[0] / dmat[0][0]);
}
for(i1 = 0, i2 = 0; i2 < rs; ++i1) {
if(i1 != ind) {
egvt[i1] = darr[i2];
++i2;
}
}
}
if(rs < size) {
Normalise(egvt);
}
for(i1 = 0; i1 < size; ++i1) {
egvts[i1][ind] = RoundToZero(egvt[i1]);
}
}
template <class T> double GaussNewton<T>::Fit(DArray &c) {
double sum = 0.;
DArray res(x_.size(), 0.);
DMatrix jac(x_.size(), t_.GetC()), dc(1, t_.GetC());
do {
//break;
ComputeResiduals(res);
ComputeJacobian(jac);
ComputeDeltas(jac, res, dc);
cerr << "Deltas: " << endl;
PrintMatrix(dc);
} while(RoundToZero(UpdateParameters(dc, c)) != 0.);
ComputeResiduals(res);
for(int i = 0; i < res.size(); ++i) {
sum += res[i] * res[i];
}
return sqrt(sum);
}
static block_t * DoWork( filter_t * p_filter, block_t * p_in_buf )
{
int i_samples = p_in_buf->i_nb_samples;
int i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
float *pf_buf = (float*)p_in_buf->p_buffer;
/* Current parameters */
filter_sys_t *p_sys = p_filter->p_sys;
/* Fetch the configurable parameters */
vlc_mutex_lock( &p_sys->lock );
float f_rms_peak = p_sys->f_rms_peak; /* RMS/peak */
float f_attack = p_sys->f_attack; /* Attack time (ms) */
float f_release = p_sys->f_release; /* Release time (ms) */
float f_threshold = p_sys->f_threshold; /* Threshold level (dB) */
float f_ratio = p_sys->f_ratio; /* Ratio (n:1) */
float f_knee = p_sys->f_knee; /* Knee radius (dB) */
float f_makeup_gain = p_sys->f_makeup_gain; /* Makeup gain (dB) */
vlc_mutex_unlock( &p_sys->lock );
/* Fetch the internal parameters */
float f_amp = p_sys->f_amp;
float *pf_as = p_sys->pf_as;
float f_env = p_sys->f_env;
float f_env_peak = p_sys->f_env_peak;
float f_env_rms = p_sys->f_env_rms;
float f_gain = p_sys->f_gain;
float f_gain_out = p_sys->f_gain_out;
rms_env *p_rms = &p_sys->rms;
float f_sum = p_sys->f_sum;
lookahead *p_la = &p_sys->la;
/* Prepare other compressor parameters */
float f_ga = f_attack < 2.0f ? 0.0f :
pf_as[Round( f_attack * 0.001f * ( A_TBL - 1 ) )];
float f_gr = pf_as[Round( f_release * 0.001f * ( A_TBL - 1 ) )];
float f_rs = ( f_ratio - 1.0f ) / f_ratio;
float f_mug = Db2Lin( f_makeup_gain, p_sys );
float f_knee_min = Db2Lin( f_threshold - f_knee, p_sys );
float f_knee_max = Db2Lin( f_threshold + f_knee, p_sys );
float f_ef_a = f_ga * 0.25f;
float f_ef_ai = 1.0f - f_ef_a;
/* Process the current buffer */
for( int i = 0; i < i_samples; i++ )
{
float f_lev_in_old, f_lev_in_new;
/* Now, compress the pre-equalized audio (ported from sc4_1882
* plugin with a few modifications) */
/* Fetch the old delayed buffer value */
f_lev_in_old = p_la->p_buf[p_la->i_pos].f_lev_in;
/* Find the peak value of current sample. This becomes the new delayed
* buffer value that replaces the old one in the lookahead array */
f_lev_in_new = fabs( pf_buf[0] );
for( int i_chan = 1; i_chan < i_channels; i_chan++ )
{
f_lev_in_new = Max( f_lev_in_new, fabs( pf_buf[i_chan] ) );
}
p_la->p_buf[p_la->i_pos].f_lev_in = f_lev_in_new;
/* Add the square of the peak value to a running sum */
f_sum += f_lev_in_new * f_lev_in_new;
/* Update the RMS envelope */
if( f_amp > f_env_rms )
{
f_env_rms = f_env_rms * f_ga + f_amp * ( 1.0f - f_ga );
}
else
{
f_env_rms = f_env_rms * f_gr + f_amp * ( 1.0f - f_gr );
}
RoundToZero( &f_env_rms );
/* Update the peak envelope */
if( f_lev_in_old > f_env_peak )
{
f_env_peak = f_env_peak * f_ga + f_lev_in_old * ( 1.0f - f_ga );
}
else
{
f_env_peak = f_env_peak * f_gr + f_lev_in_old * ( 1.0f - f_gr );
}
RoundToZero( &f_env_peak );
/* Process the RMS value and update the output gain every 4 samples */
if( ( p_sys->i_count++ & 3 ) == 3 )
{
/* Process the RMS value by placing in the mean square value, and
* reset the running sum */
f_amp = RmsEnvProcess( p_rms, f_sum * 0.25f );
f_sum = 0.0f;
if( cover_isnan( f_env_rms ) ) // sunqueen modify
{
/* This can happen sometimes, but I don't know why. */
f_env_rms = 0.0f;
}
/* Find the superposition of the RMS and peak envelopes */
f_env = LIN_INTERP( f_rms_peak, f_env_rms, f_env_peak );
/* Update the output gain */
if( f_env <= f_knee_min )
{
/* Gain below the knee (and below the threshold) */
f_gain_out = 1.0f;
}
else if( f_env < f_knee_max )
{
/* Gain within the knee */
const float f_x = -( f_threshold
- f_knee - Lin2Db( f_env, p_sys ) ) / f_knee;
f_gain_out = Db2Lin( -f_knee * f_rs * f_x * f_x * 0.25f,
p_sys );
}
else
{
/* Gain above the knee (and above the threshold) */
f_gain_out = Db2Lin( ( f_threshold - Lin2Db( f_env, p_sys ) )
* f_rs, p_sys );
}
}
/* Find the total gain */
f_gain = f_gain * f_ef_a + f_gain_out * f_ef_ai;
/* Write the resulting buffer to the output */
BufferProcess( pf_buf, i_channels, f_gain, f_mug, p_la );
pf_buf += i_channels;
}
/* Update the internal parameters */
p_sys->f_sum = f_sum;
p_sys->f_amp = f_amp;
p_sys->f_gain = f_gain;
p_sys->f_gain_out = f_gain_out;
p_sys->f_env = f_env;
p_sys->f_env_rms = f_env_rms;
p_sys->f_env_peak = f_env_peak;
return p_in_buf;
}
inline double EigenVectors::ComputeSystem(DMatrix &dmat, DArray &darr) const {
SLEQ sleq(dmat, darr);
return sleq.GetRoots(darr, RoundToZero(sleq.ComputeDeterminant()));
}
